[Freeswitch-users] WebRTC using rtp_sdes_suites=AES_CM_128_HMAC_SHA1_80

[Aqs Younas]

I would also be interested to know if you make this work.

Best Regards,

Aqs Younas

On 8 May 2018 at 22:11, Jerry Chinn <JHChinn at thenavisway.com> wrote:

> Good Day,
>
> Running FS 1.6.17 on CentOS 7.4
>
>
>
> We are running WebRTC and are required to use AEAD_AES_256_GCM_8 or
> AEAD_AES_128_GCM_8 for security.
>
> I have eliminated all of the options in the vars file except
> rtp_sdes_suites=AEAD_AES_256_GCM_8|AEAD_AES_128_GCM_8.
>
>
>
> Calls are successfully completing, however, in debug we are seeing
> AES_CM_128_HMAC_SHA1_80 as the sdes suite for srtp:dtls.
>
>
>
> 2018-05-04 22:38:30.429310 [INFO] switch_rtp.c:3185 Changing audio DTLS
> state from HANDSHAKE to SETUP
>
> 2018-05-04 22:38:30.450549 [INFO] switch_rtp.c:3094 audio Fingerprint
> Verified.
>
> 2018-05-04 22:38:30.450549 [INFO] switch_rtp.c:3908 Activating audio
> Secure RTP SEND
>
> 2018-05-04 22:38:30.450549 [DEBUG] switch_core_sqldb.c:2617 Secure Type:
> srtp:dtls:AES_CM_128_HMAC_SHA1_80
>
> 2018-05-04 22:38:30.450549 [INFO] switch_rtp.c:3886 Activating audio
> Secure RTP RECV
>
> 2018-05-04 22:38:30.450549 [INFO] switch_rtp.c:3134 Changing audio DTLS
> state from SETUP to READY
>
> 2018-05-04 22:38:30.450549 [DEBUG] switch_core_sqldb.c:2617 Secure Type:
> srtp:dtls:AES_CM_128_HMAC_SHA1_80
>
> 2018-05-04 22:38:30.450549 [DEBUG] switch_rtp.c:1885 rtcp_stats_init:
> audio ssrc[3910337773] base_seq[2433]
>
>
>
> Any ideas on how or where to change this to the desired encryption
> protocol?
>
>
>
> Jerry Chinn
>
> Telecom VoIP Specialist
>
> .
>
>
>
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